similar to: Play sounds to both caller and callee at the same time

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

Hi, I've been playing with T.38. I observed that mostly but not always, it's the "calling endpoint" that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the "standardized" or most common, way to start a T.38 session ? Shall it come from callee or

Hello, you know , when a call setup, either caller hangup first or callee hangup first , the hangupcause will set to 16(means Call Clearing Causes) My question is how could i identify whether the caller or callee hangup the phone first? Best Regards! -- Thanks for your supporting, have a nice day. Sucan

Hi all, I am setting up * for the first time, every thing is working fine, but I would like to implement an additional feature: Thus we have multilingual caller menu - I would like to play a little sound file to the callees to let them know in which language they should answer the incoming call, before I pass the caller through (caller should hear the normal ringing meanwhile). Is that possible?

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

Hi, I have been trying to enable attended transfer for callee. When the callee pressed *2, DTMF tone was heard by the caller. But when the caller pressed *2, attended transfer started. It's strange. I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built by root@router on a i686 running Linux on 2005-06-27 06:07:18". In features.conf, I have: [featuremap]

Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002,

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hi, Is there a way to "announce" a call to the callee? For instance, I've answered an incoming call, collected some info and now want to ring an extension, and make an accouncement to that extension before connecting the caller through. Thanks, Steve

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

2013/3/27 Nick Lewycky <nicholas at mxc.ca>: > charles quarra wrote: >> >> Hi, >> >> I have two functions in a module, X.foo, which is the callee, and >> Y.foo2, which calls X.foo. >> >> If i either try to run llvm::Function::eraseFromParent() on any one of >> the functions, i'll >> get this assertion error: >> >> F is

2013/3/27 Nick Lewycky <nicholas at mxc.ca>: > charles quarra wrote: >> >> Hi, >> >> I have two functions in a module, X.foo, which is the callee, and >> Y.foo2, which calls X.foo. >> >> If i either try to run llvm::Function::eraseFromParent() on any one of >> the functions, i'll >> get this assertion error: >> >> F is

Hi, I have two functions in a module, X.foo, which is the callee, and Y.foo2, which calls X.foo. If i either try to run llvm::Function::eraseFromParent() on any one of the functions, i'll get this assertion error: F is used in instruction: %"calling function" = call i32 @X.foo(i32 %read) F is used in instruction: %"calling function" = call i32 @X.foo(i32 %read) While